SPA 122 Loosing Registration
I'm having the same issue as other posts, and not seeing any solution.
Hoping that the pieces below narrow down the issue, which I'm feeling is something simple.
I have 2 numbers registered with voip.ms
When I was on DSL, no issues the registration -- rarely if ever dropped and usually recovered on its own.
Now I'm on cable and am getting this nightly drop in registration.
I have 3 devices and only the SPA122 has an issue with registration. Motorola TC55 running android and Groundwire - no issues with registration. Ipad running Softphone - no registration issues.
When I manually reboot the SPA122, it registers every time.
I tried setting NAT Keep Alive Msg: to $REGISTER from the default $NOTIFY, and it hangs in, "fixes" the registration issue ..... but I get no ring through on the SPA122, so that's not a solution.
I have the logger logging at a "Notice" level, and can send that along -- suffice to say I'm not seeing any error messages.
Tried various voip.ms servers, no change regarding the register issue.
The public IP has been the same same for weeks, so that's not changing.
All settings are as per Voip.ms wiki, have the latest firmware.
Help!
OK thank you Dan for your answer.
I bought the box from an ISP, but i don't want to use this ISP, so i can not reset the config to default settings without customization ect? With a sample.cfg file from anywhere?
Similar Messages
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SPA 122 with Polycom Equipment
I have a number of these devices, SPA 122's, set up at various customer sites with Polycom analog conference units running on a hosted telephony platform. They are all running the latest firmware. They all work great except for one problem.
A standard functionality of Polycom equipment is the ability to intercom other phones on the PBX using (* + 1 + Extension #). However, the problem that I am running into is that the IRV default access is also done by selecting (*) from the analog device. Is there a setting that I can change that would allow me to avoid this conflict? I understand I may loose the ability to configure the ATA using the IVR and will neeed to use the web management. This is fine.
Can anyone provide assistance with this?Try the following.
Update the 'dial plan' (line tab) and the 'feature dial services codes' (regional tab).
In the dial plan, remove the *xx and put in *1xx. if you plan to dial *1 and 2 or more digits afterward.
In the feature dial services codes, put in the same *1xx.
Remove the default settings for Secure call in the Vertical Service Activation Codes, since those * codes start with *1. -
We just upgraded SPA 122 to version 1..1.1 Ver 011. After the upgrade SPA cannot register to the PBX to which it was registered before. PBX doesn't even show any registration attempts by it. From the SPA we can ping the domain to which it is trying to register. Confused.
Thank you Patrick,
Here is the result of the capture after these instructions. Interestingly in this log I do not even see an attempt to register line 1. We have configured line 1 to register to localpbx and line 2 to register to babytel trunk...
++++ retry query scaps
+++ need tftp addr..
+++ send scaps discovery query
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-d519873d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12899 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
RSE_DEBUG: getting alternate from domain:nat.babytel.ca
CC_eventProc(), event: CC_EV_SIG_REGISTER_FAILED(0x3B), lid: 1, par: 0, par2: (n
il)
AUD_ccEventProc: event 59 vid 1 par 0x0 par2 0x0
SLIC_stopRing
SLIC_stopTone
[1]RegFail. Retry in 30
[rse_refresh_addr_list] query nat.babytel.ca block 0
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-adf86974
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12901 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
RSE_DEBUG: getting alternate from domain:nat.babytel.ca
CC_eventProc(), event: CC_EV_SIG_REGISTER_FAILED(0x3B), lid: 1, par: 0, par2: (n
il)
AUD_ccEventProc: event 59 vid 1 par 0x0 par2 0x0
SLIC_stopRing
SLIC_stopTone
[1]RegFail. Retry in 30
++++ retry query scaps
+++ need tftp addr..
+++ send scaps discovery query
[rse_refresh_addr_list] query nat.babytel.ca block 0
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[1]->204.101.5.67:5060(658)
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 10.3.12.160:5061;branch=z9hG4bK-8a2f808d
From: "babytel" [email protected]>;tag=f55060d797aa79do1
To: "babytel" [email protected]>
Call-ID: [email protected]
CSeq: 12903 REGISTER
Max-Forwards: 70
Contact: "babytel" ;expires=3600
P-Station-Name: ;mac=ccef485c283e; sn=CBT154400J2
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
P-Station-Name: ;mac=ccef485c283e; display=""; sn=CBT154400J2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
RSE_DEBUG: getting alternate from domain:nat.babytel.ca
CC_eventProc(), event: CC_EV_SIG_REGISTER_FAILED(0x3B), lid: 1, par: 0, par2: (n
il)
AUD_ccEventProc: event 59 vid 1 par 0x0 par2 0x0
SLIC_stopRing
SLIC_stopTone
[1]RegFail. Retry in 30 -
We provision our SPA 122 ATAs via a provisioning server. We have been using this method successfully to register our ATAs with our voice server. For the past two days we have been unable to register any 122 ATAs. The ATAs are pulling the configuration, as well as the upgrade file (spa122-1-2-1-004), however they are not registering. When we use the same method for the old SPA 2102 ATAs, the configuration and the upgrade file are downloaded and the ATA registers. Should also note that the SPA 122 is also receiving an IP address and we can browse the internet through the device. We would appreciate any advice you can give.
Regards,
Nikolas RockWe need to look at the debug log to see if the phone is sending out any registration messages.
https://supportforums.cisco.com/docs/DOC-9862 for info on setting up debug log.
If it's not sending out messages, we need to see the configuration. Thanks. -
SPA 122 remove restricted access domains
Hello
My spa 122 has a customization profile active and i want to remove it. But i don't know how to do it. Firmeware Update, Factory reset, disabling provisioning don't help.
I tried to create a sample config file with the Cisco Profile Compiler but no luck, the spa 122 tell me everytime Restore failed...
All i want is to remove the settings from the customization, like "restricted access domains" ect.
Can anyone help me?
Thanks
mfg
PascalOK thank you Dan for your answer.
I bought the box from an ISP, but i don't want to use this ISP, so i can not reset the config to default settings without customization ect? With a sample.cfg file from anywhere? -
Well what can I say ? we purchased a few 122's for testing & product evalation and with the latest firmware SPA1xx_1.1.0_011.bin about a week after flashing it, there is now static sounds and buzzing coming from spa-122 ata, once I swapped out the spa-122 for a good old pap2t or my 2102 units works like a charm sounds great and no issues period, Has anyone seen or heard of an issues reguarding dialtone static/buzzing sounds?
ThanksFYI, just got today another inacceptable response from Cisco, honestly I'm very frustrated with this new SPA122 that has been release to public without proper QA, that replacement of the SPA2102 should NOT has been release to public, I have a OPEN CASE since September 2012 > 9 months open ticket, troubleshooting hell with them, and they can't even offer me a replacement, upgrade to working product, offer alternative... Warranty is about to expire, then I'll be stuck with a broken, untested and useless device. What are our rights?!? I may sound very ULTRA frustrated, well I AM. Cisco customer service suck with VOIP product. I've been in network engineering for > 22 years, never experience that low quality support with Cisco and other vendors... I can't even escalate the issue to upper, they are just not cooperative. Not only EVERY owner of an SPA122 are having unstable issues, well too bad for us for purchasing that item, we are alone without support and endless efforts on reseting the bad not working devices...
Patrick Born, can you do something for me (us), do you think it's acceptable that kind of services?
Response of today: That's quality and service...
----- Message d'origine -----
De : Alex xxxxxx [mailto:[email protected]]
Envoyé : Thursday, April 25, 2013 05:26 PM Eastern Standard Time
À : me
Objet : Re: spa122 - High pitch noise with dial tone when phone is off hook
Greetings,
The following is a case status update courtesy notice.
The issue you reported remains open with Engineering & Development teams.
This issue may be addressed in a forthcoming Maintenance Release firmware,
however there is no ETA for this release. We will continue to monitor
Engineering & Development team progress and notify you as soon as any
updated information becomes available. Please let us know if you have any
questions.
Kind regards, -
SPA 3102: loosing connection intermittently
Hi,
I am using SPA 3102 and D-Link wireless router together. Connection is Cable Modem to SPA 3102 to D-Link DIR-615.
Both the phone and wirless works fine in this set up. The main issue is when I use utorrent, SPA 3102 keeps resetting (loosing connection intermittently). All the three lights goes of for a second and then comes back. End result is, I have to keep resetting the routers (power off for 10 secs and switch on). This is really annoying, I have to keep doing it atleast 3 or 4 times a day. I guess the problem might be with the heavy downloading/uploading with utorrent. Any way out ? Help appreciated.
ajOoops, sorry My first post was a mistake. Here is my question:
I have modem connection to a remote site.This connection must go from wire to Ethernet. The modems are 2-wire, not dial-up, and are embedded in a device, so there is no access to the serial port of the device.
I have two SPA3102. On the remote site the modem is always connected to the FXS port. This port is configured with SAS Enable=Yes and SAS Inbound RTP Sink = $IP. On the local site when modem is connected the conection is automatically established. With Wireshark I can see that RTP audio stream is flowing in both directions but signal can not be heared on the server side. What should I do to hear the inbound stream? Is it possible with SPA3102?
First I tried with SPA112 as SAS on remote site. The signal passes in both directions and can be heared on SPA112 port. Same settings. But SPA112 causes delay of 140ms and timing is critical for my devices. SPA3102 has only 80ms delay with RTP Packet Size = 0.02 and modems work with two SPA3102 configured as hot line. The problem with hot line is that in case of connection fail I have to go to the remote site to disconnect the modem and connect it again. Not very convinient.
Thanks
Svilen -
SPA 122 stops working with both PHONE 1 and 2 lights flashing
Hi,
We have hundreds of units deployed and we start having complaints that the line isn't working anymore while the unit does register with our server. Any calls going to the ATA gets a USER_BUSY condition like if a line was offhook but that's not the case.
Anyone had this condition? I am running 1.3.1(003).
One common thing so far is the fact those users do faxing but we ruled out config since we enable T.38 on all of them regardless.
Since the unit registers, we can't rely on the REGISTRATION too much so would like a way to figure out if the unit is healthy or not.
Thanks,
BenoitPlease update to 1.3.2(014) and see if this resolves the issue.
http://software.cisco.com/download/release.html?mdfid=283998793&softwareid=282463187&release=1.3.2&relind=AVAILABLE&rellifecycle=&reltype=latest -
SPA 122 returns 180 ringing after 10 seconds
Hi Folks,
I have this weird issue where the SPA takes 9.7 seconds (as per looking in the tcpdump) after receiving the INVITE to return the 180 Ringing. But it returns the 100 Trying after 0.06 seconds so I know it got the INVITE.
t0 => INVITE sent to SPA
t0+0.06 => SPA returns 100 Trying
t0+9.7 => SPA returns 180 Ringing
The phone only ring at the time the 180 is returned. It is a standard installation (as with my multiple other SPA devices out there) and I already had my customer restart the cable modem, the router and the ATA.
Any hints as to what I should check or any debug I should collect would be appreciated. I have a strong background in SIP and VoIP but this type of ATA is fairly new and I am sure it is something within the ATA that is not properly working.
This ATA uses NAT and runs FW 1.3.1(003).
Thanks much!
BenoitHi Benoit,
Use the https://supportforums.cisco.com/docs/DOC-26697 document to enable debug on your ATA and use Wireshark or a syslog server to collect the resulting output. The output may give us some clues as to the cause of the delay.
What you're describing is indeed pretty weird. I've just tested in my lab using a factory-defaulted SPA122 running 1.3.1(003) which I registered to two different ITSPs. An inbound call to the SPA122 FXS port results in a 100 TRYING followed 0.12 seconds later by the 180 RINGING. I see no delay between the 100 and 180.
It may be worth factory-defaulting the ATA in case its configuration has been to something unknown.
Regards,
Patrick---
Use this reference document to locate SPA ATA resources -
Spa 122 inbound problem "Number called not in service
Hello When i call my home # for first few tries i get number is not service then call goes thru .is there is any setting in SPa 122
Set up the debug log so you can see if the proxy is forwarding the call to the spa122.
Info on setting up debug log at https://supportforums.cisco.com/docs/DOC-9862
After setting this up, make a call and see if the call shows up in the log file. If it doesn't, then you'll know that the proxy isn't forwarding the call to the unit.
You can also check if the unit is/isn't registered at the time of the call. Check the info tab of the web UI. -
SPA 122 - Loopback capabilities
Hi,
I am looking at ways to test audio quality when a customer reports audio issue. The PBX can echo audio and I would like to see if the ATA could do the same so that I could originate a tone from the PBX to an ATA in loopback mode so I can listen to it and check the quality I recieve.
Ideally would be to have a command (like sending a NOTIFY to the unit) sent to the ATA to put the FXS in loopback. Or have a some physical cable we could ask our customer to plug on PHONE1 and PHONE2 so I could call one FXS and answer that call on the second line ...
Anyone has ideas about that?
Thanks,
BenoitYes, Linksys devices are "loopback call" capable. They can be either loopback stream source as well as loopback deflector. So you can initiate loopback call to end-user's ATA device with no user intervention. The called device will transmits the audio packets that it receives back to the transmitter/receiver instead of transmitting the data sampled on attached analog telephone. Unfortunatelly, the loopback call feature is not documented by Cisco (as far as I know). On the other side, it seems that RFC 6849 has been created with the Linksys's implementation in the mind and Cisco's employee is co-author of such RFC. So it may be relevant and may help to understand the feature details despite Linksys implementation may deviate from it a lot.
Following informations are known to me.
No, such feature is not triggered by a NOTIFY message. The loopback call is negotiated in SDP. See catched SDP bellow:
Standard call
Loopback call
(source/media mode)
m=audio 16532 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=audio 16530 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=loopback:rtp-media-loopback
a=loopback-source:8
m=audio 16530 RTP/AVP 113
a=loopback:rtp-start-loopback
a=rtpmap:113 PCMA/8000
In packet media loopback type, the loopback:rtp-pkt-loopback is used instead of loopback:rtp-media-loopback
In mirror media loopback mode, the loopback-mirror:8 is used instead of loopback-source:8 and m=...113 media descriptor is not present at all.
The id 113 is id of RTP-Start-Loopback Dynamic Payload as configured in phone setup (113 is default value). The id 8 in loopback-source/loopback-mirror is id of RTP-Start-Loopback Codec as configured in phone setup (here 8 = PCMA).
Following setup options are related to loopback call feature:
Media Loopback Code
The star code used for enabling media loopback on the phone.
The default i *03.
Accept Media Loopback Request
Controls how to handle incoming requests for loopback operation. Choices are: Never, Automatic, and Manual,
where:
never — never accepts loopback calls; reply 486 to the caller
automatic — automatically accepts the call without ringing
manual — rings the phone first, and the call must be picked up manually before loopback starts
The default is Automatic.
Media Loopback Mode
The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror.
Default is Source.
Note that if the ATA device answers the call, the mode is determined by the caller.
Media Loopback Type
The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.
Default is Media.
Note that if the ATA device answers the call, then the loopback type is determined by the caller (the ATA device always picks the first loopback type in the offer if it contains multiple types.)
ENCAP RTP Dynamic Payload
The dynamic payload value (96 – 127) used for the encapsulating RTP packets when offering the SDP to loopback packets. This setting is used if the SPA is the offerer of the SDP. Otherwise, the value is decided by the peer.
The default value is 112.
RTP-Start-Loopback Dynamic Payload
The dynamic payload value (96 – 127) used by the mirror in the self-generated RTP packets before receiving any RTP packets from the source. This setting is used only when the SPA is acting as the loopback source. Otherwise, the value is decided by the peer.
This value must be different from any of the dynamic payload values that might be used by the source and the mirror (including the encaprtp payload type). This is necessary so that the source can easily tell when the mirror has switched from sending self-generated RTP packets to sending loopback packets.
The default value is 113.
RTP-Start-Loopback Codec
The actual codec corresponding to RTP-Start-Loopback Dynamic Payload, whose codec name is used in the rtpmap attribute for the for the mirror self-generated RTP audio stream, prior to receiving any RTP packets from the source. -
Newly fetched settings getting reset all the time (provisioning of SPA 122)
The initial configuration works and is pretty much a sample file that sets some defaults and resets the provisioning rule to fetch an encrypted file with this information:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<flat-profile xmlns="http://www.sipura.net/xsd/SPA122" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://www.sipura.net/xsd/SPA122 http://www.sipura.net/xsd/SPA122/SPA122-1-3-2-014.xsd">
<Resync_Periodic ua="na">3600</Resync_Periodic>
<Resync_Error_Retry_Delay ua="na">3600</Resync_Error_Retry_Delay>
<Display_Name_1_ ua="na">number</Display_Name_1_>
<User_ID_1_ ua="na">number</User_ID_1_>
<Password_1_ ua="na">password</Password_1_>
<Use_Auth_ID_1_ ua="na">No</Use_Auth_ID_1_>
<Auth_ID_1_ ua="na"></Auth_ID_1_>
<Resident_Online_Number_1_ ua="na"></Resident_Online_Number_1_>
<SIP_URI_1_ ua="na"></SIP_URI_1_>
<Display_Name_2_ ua="na"></Display_Name_2_>
<User_ID_2_ ua="na"></User_ID_2_>
<Password_2_ ua="na"></Password_2_>
<Use_Auth_ID_2_ ua="na">No</Use_Auth_ID_2_>
<Auth_ID_2_ ua="na"></Auth_ID_2_>
<Resident_Online_Number_2_ ua="na"></Resident_Online_Number_2_>
<SIP_URI_2_ ua="na"></SIP_URI_2_>
</flat-profile>
I can see in /admin/config.xml on the adapter that it downloads and loads the settings for only a couple of seconds, but then quickly reverts back to the settings it had before downloading these - what is going on??
Regards,
MikaelSeems my SPA122 is broken, the same procedure works when I tried another one. Yawn!
-
SPA 122 initial config works, but cannot complete outgoing calls
I configured the ATA exactly step by step anbd I did follow the guidelines of my voip provider (voip.ms).
I see I am registered, everything works well, I get the dial tone, I can dial but then nothing happens. I checked the dial plan and tried several times to reset the unit and configure again from scratch...what am I doing wrong?Hello Nathan,
The configuration of the hunt group is not correct.
You cannot use call forward all under extension in hunt group because it is ignored this is the reason why the call flow is not working.
This is one of the restrictions of the hunt groups, please check the following link.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#wp1117433
You need to set the call flow in another way.
HTH,
Alex
*Please rate helpful posts -
SIP Registration is blocked by BT or Home Hub
Hi all,
I have a problem, between 7.02PM and 11.02PM my SIP Softphone looses registration - every night. I have isolated the issue to my BT Infinity connection as when I close my WIFI and use 3G the SIP telephone registers sucessfully.
I 'think' it's either my Home Hub 3 or BT themselves blocking SIP Registration
Anyone seen this before?
Regards
MattIs your SIP telephone connected by a cable, or does it use WiFi?
If it uses WiFi, then you may be getting wireless interference, and you would need to select a different wireless channel.
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones. -
Transcoding Sessions unregistered with CUCM from standby gateway of CUBE with HA usingHSRP
I have 2 C2921 routers working fine untill I enbale CUBE with HA. After configureing HSRP on ethernet interface, the transcoding and conferencing resources were unregistered on HSRP standby router even though I bind the sccp ccm group to physical interface.
Raised Cisco TAC, but they also could not solve yet. Cisco TAC recommanded to have loopback or another interface.
I configure gi02/ without HSRP configuration, but still the transcoding and conferencing resources are not getting registered. Cisco TAC is still analysing the logs.
I am hoping I get resolution here. Configuration of standby router is below.
Building configuration...
Current configuration : 13985 bytes
! Last configuration change at 15:07:25 BST Fri Aug 1 2014
! NVRAM config last updated at 15:07:25 BST Fri Aug 1 2014
version 15.4
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
service internal
service sequence-numbers
hostname CHN-RT-VG01
boot-start-marker
boot system flash:c2900-universalk9_npe-mz.SPA.154-3.M.bin
boot system flash:c2900-universalk9_npe-mz.SPA.154-2.T1.bin
boot-end-marker
! card type command needed for slot/vwic-slot 0/0
card type e1 0 1
card type e1 0 2
logging queue-limit 10000
logging buffered 10000000
logging rate-limit 10000
no logging console
enable secret 4 XkK1t85uKpzHay4O0x8hP0rt1uO7UwNlcWBLwLAsn3Y
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 10.215.8.148
remote-port 5000
remote-ip 10.215.8.149
--More-- no aaa new-model
clock timezone BST 0 0
clock summer-time BST date Mar 28 1993 0:00 Oct 27 2035 23:59
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-select 1 E1 0/1/0
network-clock-select 2 E1 0/1/1
network-clock-select 3 E1 0/2/0
no ip domain lookup
ip domain name DILFLPROD.CO.UK
--More-- ip cef
ipv6 multicast rpf use-bgp
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-4ess
cts logging verbose
crypto pki trustpoint TP-self-signed-3464013556
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3464013556
revocation-check none
rsakeypair TP-self-signed-3464013556
crypto pki certificate chain TP-self-signed-3464013556
certificate self-signed 01
3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 33343634 30313335 3536301E 170D3132 31313232 30353530
30345A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 34363430
31333535 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100FD06 30324087 5D131745 446B6933 963E32DB 4B3F78D3 C2627F7B A68792EA
0686B7C1 93B66C1A 2287DD72 26AC10BE F6B5DE89 CEF9C800 836DAD25 4A32FC52
99A65E45 FAD97919 4BD2CFC8 136EB9AC F7F21045 0A930247 0E72CE1B 1C00D1BD
59B83BED 73639AA5 C78A657B EC55F15B 5287703C 3ED94E47 492DFAD0 89934B27
5CD10203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603
551D2304 18301680 146F6961 3C46FDE7 C105ADBF 5C07A675 7F7B5828 E1301D06
03551D0E 04160414 6F69613C 46FDE7C1 05ADBF5C 07A6757F 7B5828E1 300D0609
2A864886 F70D0101 05050003 8181005E 509EACC9 67205643 133DD745 5A6E7C82
7AAE0766 C68C215B 6222A86F A08AC77D 1030664E F77F6CFB CF021C94 BC5FB190
FEA96EE9 5A502DC6 D4407467 9662683E CFDC1779 4016A9A0 32EF415D 6E21DF53
D710D173 7BFC300A FDEE54D8 36BBED28 05A6A752 652F2550 E6BC5896 D4EC222A
C82C1B2A 4FEF6ED3 44DE109E DD796E
--More-- quit
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice service voip
mode border-element
allow-connections sip to sip
redundancy
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
early-offer forced
midcall-signaling passthru
g729 annexb-all
voice translation-rule 100
rule 1 /^44845..\(.....\)/ /\1/
rule 3 /^4411...\(....\)/ /2\1/
voice translation-profile LiveOpsInbound
translate called 100
voice translation-profile OutboundtoKolDDI
translate called 1
--More-- !
application
global
service alternate Default
license udi pid CISCO2921/K9 sn FCZ164760NP
hw-module pvdm 0/0
hw-module pvdm 0/1
username controller privilege 15 password 7 050F0F03284B4B070D04
username voiceadmin privilege 15 password 7 1514190501242F37243A3327
username shaums privilege 15 password 7 151602000D2D2E2A3C32
username 745162 privilege 15 password 7 08254542001E0019060A
username 256108 privilege 15 password 7 0124030858040B0A70
redundancy inter-device
scheme standby SB
redundancy
no keepalive-enable
notification-timer 60000
controller E1 0/1/0
pri-group timeslots 1-31 service mgcp
controller E1 0/1/1
pri-group timeslots 1-31 service mgcp
controller E1 0/2/0
pri-group timeslots 1-31 service mgcp
controller E1 0/2/1
pri-group timeslots 1-31 service mgcp
track 1 interface GigabitEthernet0/0 line-protocol
track 2 interface GigabitEthernet0/1 line-protocol
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description **Inside***
ip address 10.215.8.132 255.255.255.240
standby delay minimum 30 reload 60
standby version 2
standby 1 ip 10.215.8.135
standby 1 priority 50
standby 1 preempt
standby 1 name SB
standby 1 track 2 decrement 10
duplex auto
speed auto
interface GigabitEthernet0/1
description **Outside***
ip address 10.215.8.148 255.255.255.240
standby delay minimum 30 reload 60
standby version 2
standby 2 ip 10.215.8.150
standby 2 priority 50
standby 2 preempt
standby 2 track 1 decrement 10
duplex auto
speed auto
media-type rj45
--More-- !
interface GigabitEthernet0/2
ip address 10.215.8.164 255.255.255.240
duplex full
speed 1000
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
interface Serial0/1/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
interface Serial0/2/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
ip forward-protocol nd
--More-- ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip rtcp report interval 3000
ip route 0.0.0.0 0.0.0.0 10.215.8.129
ip sla auto discovery
ip sla 40001
udp-jitter 10.215.191.3 17000 source-ip 10.215.8.132 codec g729a codec-numpackets 100
tos 184
owner SW.IpSla.CHVISM0210.SolarWindsOrion
frequency 300
timeout 180000
threshold 1000
ip sla schedule 40001 life forever start-time now
ip sla 40003
udp-jitter 10.215.221.131 17000 source-ip 10.215.8.132 codec g729a codec-numpackets 100
tos 184
owner SW.IpSla.CHVISM0210.SolarWindsOrion
frequency 300
timeout 180000
threshold 1000
ip sla schedule 40003 life forever start-time now
no logging trap
snmp-server community m&9C4rd4L%mw RO 10
snmp-server community m&9C4rd4L%m RW 10
snmp-server enable traps isdn call-information
snmp-server enable traps isdn layer2
snmp-server enable traps isdn chan-not-avail
snmp-server enable traps isdn ietf
snmp-server host 10.215.10.10 version 2c m&9C4rd4L%mw
snmp-server host 10.215.232.202 version 2c m&9C4rd4L%mw
tftp-server flash0:SCCP42.9-1-1SR1S.loads
tftp-server flash0:apps42.9-1-1TH1-16.sbn
tftp-server flash0:cnu42.9-1-1TH1-16.sbn
--More-- tftp-server flash0:cvm42sccp.9-1-1TH1-16.sbn
tftp-server flash0:dsp42.9-1-1TH1-16.sbn
tftp-server flash0:jar42sccp.9-1-1TH1-16.sbn
tftp-server flash0:term42.default.loads
tftp-server flash0:term62.default.loads
tftp-server flash0:/c2600-ipvoicek9-mz.124-25d.bin
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
voice-port 0/1/0:15
voice-port 0/2/0:15
voice-port 0/1/1:15
voice-port 0/2/1:15
mgcp
mgcp call-agent 10.215.8.7 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface GigabitEthernet0/0
mgcp bind media source-interface GigabitEthernet0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
--More-- mgcp behavior comedia-sdp-force disable
mgcp profile default
sccp local GigabitEthernet0/2
sccp ccm 10.215.8.7 identifier 1 priority 1 version 7.0
sccp ccm 10.215.8.6 identifier 2 priority 2 version 7.0
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register CFBCHEVG1
associate profile 1 register XCODERCHEVG1
ccm-manager music-on-hold
ccm-manager fallback-mgcp
ccm-manager redundant-host 10.215.8.6
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager config server 10.215.8.6
ccm-manager config
dspfarm profile 1 transcode
codec g729r8
codec ilbc
codec pass-through
codec g722-64
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 70
associate application SCCP
dspfarm profile 2 conference
codec g729br8
codec g729r8
codec g729abr8
--More-- codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
codec ilbc
maximum sessions 10
associate application SCCP
dial-peer voice 1 pots
description **Incoming Dial Peer**
incoming called-number .
direct-inward-dial
dial-peer voice 2 pots
description **Outbound Dialpeer**
translation-profile outgoing OutboundtoKolDDI
destination-pattern 02083917600
incoming called-number .
port 0/1/1:15
dial-peer voice 3 pots
description **Outbound Dialpeer**
translation-profile outgoing OutboundtoKolDDI
destination-pattern 02083917600
incoming called-number .
port 0/2/0:15
dial-peer voice 4 pots
description **Outbound Dialpeer**
translation-profile outgoing OutboundtoKolDDI
destination-pattern 02083917600
incoming called-number .
port 0/2/1:15
dial-peer voice 100 voip
description to-DorkingCUCM
translation-profile outgoing LiveOpsInbound
destination-pattern 44..........
session protocol sipv2
session target ipv4:10.156.125.2
--More-- incoming called-number .
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 200 voip
description to-LiveOpsCCC
preference 1
destination-pattern .T
session protocol sipv2
session target ipv4:x.x.x.x
incoming called-number 44..........
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 300 voip
description to-LiveOpsCCC
preference 2
destination-pattern .T
session protocol sipv2
session target ipv4:x.x.x.x
incoming called-number 44..........
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 101 voip
description to-ChessingtonCUCM
translation-profile outgoing LiveOpsInbound
preference 1
destination-pattern 44..........
session protocol sipv2
session target ipv4:10.215.8.7
--More-- incoming called-number 40008
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 102 voip
description to-ChessingtonCUCM
translation-profile outgoing LiveOpsInbound
preference 2
destination-pattern 44..........
session protocol sipv2
session target ipv4:10.215.8.6
incoming called-number 40008
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 103 voip
description to-DorkingCUCM
preference 1
shutdown
destination-pattern 25544
session protocol sipv2
session target ipv4:10.156.125.2
incoming called-number .
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 104 voip
description to-ChessingtonCUCM
translation-profile outgoing LiveOpsInbound
preference 1
shutdown
--More-- destination-pattern 40008
session protocol sipv2
session target ipv4:10.215.8.7
incoming called-number .
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711ulaw
no vad
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
gatekeeper
shutdown
banner login ^CC
"This system and components thereof is the sole and exclusive property of Diligenta and is intended solely for the usage of its authorized administrators. Unauthorized access or use will attract appropriate legal action.
Access would be bound by Diligenta policies and could be monitored. Do not use this system, if the terms are not acceptable."
^C
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
privilege level 15
login local
--More-- transport input ssh
line vty 5 15
privilege level 15
login local
transport input ssh
scheduler allocate 20000 1000
endI don't have an answer for you but would like to add a note. I was initially configuring and troubleshooting some things on a HA cube pair recently. I was using external DNS for some dial-peer session target lookup and noticed the non-active CUBE could not lookup DNS. When the non-active CUBE became active it could all of a sudden resolve DNS. So I am speculating that something to do with the HA configuration is disallowing communication or bindings preventing routing to the rest of the network from the non-active CUBE. I ended up putting local host records on the router to make me feel better. I am guessing whatever is causing that might be related to the reason your SCCP is loosing registration on the non-active CUBE.
Jaime says what you are trying to do is not supported anyway. I would like a a little clarification on that but what I believe to be supported is if you need transcoding or mtp resources for this CUBE only (Not registered to UCM) then LTI is a good option.
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115018-configure-cube-lti.html
Hope any of this helps. I am really commenting so I can track any updates to this thread. :)
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