Re: [Asterisk-Users] New Firefly version
adam - can the g729.dll be downloaded somewhere - is this still required for g.729 support? Regards, Steven Thomas jo [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 31/05/2004 09:19 PM Please respond to asterisk-users To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] New Firefly version Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pulver WiSIP Dual Line and Hold?
Hi, I have received my WiSIP phone - works well for basic functions of call answer and hang-up! Does anyone know how to enable Dual line support, Hold and Transfer functions with this phone via Asterisk. Thanks, Regards, Steven Thomas
[Asterisk-Users] BCM Wireless SIP Phone
Hi, Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm Regards, Steven Thomas Network Integration Services IBM Australia Ph: 0404 099 262 NH011, IBM Centre, 601 Pacific Hwy, St Leonards, 2065.
Re: [Asterisk-Users] Cisco Gateway Integration
yes. Cisco 2612 Router with 2 x FXO's and 2 x FXS's. Works well using H323, and gnugk. Steve. Bruce Hedreen [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 15/12/2003 09:57 AM Please respond to asterisk-users To: [EMAIL PROTECTED] cc: Subject: [Asterisk-Users] Cisco Gateway Integration Has anyone succesfully integrated * with a cisco voice gateway ?
[Asterisk-Users] RTP Codec Error(s) - Is there really a solution for this or these?
Hi All, I have add the below error ever since installing and running with * for the past 6 months. It only occurs on calls from * to a H.323 gateway. I am using chan_h323. I have searched HI and LOW for a solution within the archives and elsewhere. When the error presents on the console - there is a millisecond pause in the active voice call . . . . NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received Thanks in desparation for any ideas Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its
[Asterisk-Users] Call logging In and Out
Is it possible to log the CallerID of an inbound call including the time to a log / text file? Also the same for outbound? ie., dialed number and time? Thanks. Regards, Steven Thomas
Re: [Asterisk-Users] delay problem in h323
I assume it manages the signal part of the RTP stream but not the RTP voice stream at the codec level? Maybe someone else can comment on the translation methodologies within Asterisk? Regards, Steven Thomas andrea [EMAIL PROTECTED] Sent by: To: [EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 04:45 PM Please respond to asterisk-users thanks, I'll try. Question: asterisk always manages RTP flow also with chan_h323? Andrea Steven Thomas wrote: Hi, I use Asterisk as a SIP - H323 translator without any issues after switching to chan_h323. My environment is: SIP (7960) - Asterisk - GnuGK (h323) - Cisco 2600 H323 Gateway to PSTN. This works well without the CPU load seen with oh323. The call control also seems far better using chan_h323. I have no delay either. I use a smaller box: PII 200, 64Mb RAM. RedHat 9. Only 4 handsets - 2 SIP IP 7960's, 2 Analog H323 via the Cisco router FXS ports. I also have configured Asterisk on another site to act as a H323 gateway for PSTN calls into a Cisco Call Manager via gnuGK - H323 also. I would suggest trying chan_h323 as an alternative. Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its andy [EMAIL PROTECTED] Sent by: To: [EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 08:24 AM Please respond to asterisk-users yes, I agree with you. I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk decode and then code again RTP flows. This function requires 5-7% CPU On my test-box (Linux rh 7.3 on P3 600 GHz). This solution don't scale without dedicated HW, I think! Another problem is codec supported: ok for G.711, G.729. I don't know for GSM BUT: what about video codec? what about proprietary codec or ciphered codec? Do you have any suggestion on how I can manage this with asterisk? I'm very interested into asterisk as sip-to-h323 translator. Thanks Andrea Quoting Steven Thomas [EMAIL PROTECTED]: The only way I was able to solve my delay issue with Chan_oh323 was to switch to Chan_h323. Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation. Checking the CPU stats on asterisk during the call - confirms that the RTP stream was somehow routing through asterisk - not sure why! Regards, Steven Thomas andrea [EMAIL PROTECTED] Sent by: To: [EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 12:45 AM Please respond to asterisk-users Hi all, is it possible to disable RTP routing through asterisk? RTP routing is a very nice feature but, I think its important also to disable it in some cases (e. g. in a LAN). Do you have any suggestion? Andrea Rattana BIV wrote: Hi, I have a delay between two H323. Netmeeting1 - || | gnuGK
Re: [Asterisk-Users] delay problem in h323
The only way I was able to solve my delay issue with Chan_oh323 was to switch to Chan_h323. Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation. Checking the CPU stats on asterisk during the call - confirms that the RTP stream was somehow routing through asterisk - not sure why! Regards, Steven Thomas andrea [EMAIL PROTECTED] Sent by: To: [EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 12:45 AM Please respond to asterisk-users Hi all, is it possible to disable RTP routing through asterisk? RTP routing is a very nice feature but, I think its important also to disable it in some cases (e. g. in a LAN). Do you have any suggestion? Andrea Rattana BIV wrote: Hi, I have a delay between two H323. Netmeeting1 - || | gnuGK | --- [asterisk-oh323] | Asterisk | Netmeeting2 --|| Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2 receive the voice without delay. But in the other way I have 3 secondes delay. In oh323.conf I set jittermin and jittermax to 20, the ipTos=lowdelay. I try to find where I can delete the delay. Does anyone have a tip ? Best Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Ë^®+$R²f¢)à+-Ë^®+$R²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
Re: [Asterisk-Users] delay problem in h323
Hi, I use Asterisk as a SIP - H323 translator without any issues after switching to chan_h323. My environment is: SIP (7960) - Asterisk - GnuGK (h323) - Cisco 2600 H323 Gateway to PSTN. This works well without the CPU load seen with oh323. The call control also seems far better using chan_h323. I have no delay either. I use a smaller box: PII 200, 64Mb RAM. RedHat 9. Only 4 handsets - 2 SIP IP 7960's, 2 Analog H323 via the Cisco router FXS ports. I also have configured Asterisk on another site to act as a H323 gateway for PSTN calls into a Cisco Call Manager via gnuGK - H323 also. I would suggest trying chan_h323 as an alternative. Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its andy [EMAIL PROTECTED] Sent by: To:[EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 08:24 AM Please respond to asterisk-users yes, I agree with you. I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk decode and then code again RTP flows. This function requires 5-7% CPU On my test-box (Linux rh 7.3 on P3 600 GHz). This solution don't scale without dedicated HW, I think! Another problem is codec supported: ok for G.711, G.729. I don't know for GSM BUT: what about video codec? what about proprietary codec or ciphered codec? Do you have any suggestion on how I can manage this with asterisk? I'm very interested into asterisk as sip-to-h323 translator. Thanks Andrea Quoting Steven Thomas [EMAIL PROTECTED]: The only way I was able to solve my delay issue with Chan_oh323 was to switch to Chan_h323. Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation. Checking the CPU stats on asterisk during the call - confirms that the RTP stream was somehow routing through asterisk - not sure why! Regards, Steven Thomas andrea [EMAIL PROTECTED] Sent by: To: [EMAIL PROTECTED] [EMAIL PROTECTED]cc: .digium.com Subject: Re: [Asterisk-Users] delay problem in h323 10-09-03 12:45 AM Please respond to asterisk-users Hi all, is it possible to disable RTP routing through asterisk? RTP routing is a very nice feature but, I think its important also to disable it in some cases (e. g. in a LAN). Do you have any suggestion? Andrea Rattana BIV wrote: Hi, I have a delay between two H323. Netmeeting1 - || | gnuGK | --- [asterisk-oh323] | Asterisk | Netmeeting2 --|| Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2 receive the voice without delay. But in the other way I have 3 secondes delay. In oh323.conf I set jittermin and jittermax to 20, the ipTos=lowdelay. I try to find where I can delete the delay. Does anyone have a tip ? Best Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ^+$Rf)+- ^+$RXb+rXb+r+-w- z --- This mail sent through CSP Webmail System ___ Asterisk-Users
[Asterisk-Users] CallerID through the GnuGK - does this work?
Hi - can anyone confirm or deny that CallerID works through (passes through) the GnuGK? ie., X100P - Asterisk - GnuGK - Gateway Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 support for phone numbers via gateway?
Does chan_h323 support phone number calling via a gateway? ie., something like calling 5000 forwarded to: exten = 5000,1,Dial(h323/[EMAIL PROTECTED]) if so - what format should the exten be in? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway
Continuing my problems with h323. I think I am getting closer. SJPhone works direct to the gateway - calls and answers fine on the pstn. So the gateway is working. Inbound calls from PSTN = Gateway = Asterisk = Phone work great! Outbound from Asterisk = Gateway = PSTN still remains a problem. The debug stuff on the gateway receives the call signal from asterisk - but does not receive the number to call - its errors with callID is -1 (nothing to call) Any ideas for the correct format to use within extenensions.conf for outbound phone number via chan_h323 and a gateway? h323 works fine if it is just an IP address that it is calling, ie, a softphone. Thanks for your help Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 one way audio
Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No voice call from H.323-phone to SIP-phone
Hi, Did anyone have any comments on the below problem - or did you (shong ching) manage to solve this? I have the same issue - any assistance would be great. Thanks. Regards, Steven Thomas shong ching [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: [Asterisk-Users] No voice call from H.323-phone to SIP-phone .digium.com 12-08-03 05:43 PM Please respond to asterisk-users Hi lists, I am trying to connect SIP Phone and H323 Phone. I can call to from SIP-Phone to H323 with clear voice. But I can't hear the voice calling from H323-phone to SIP-phone. The ring and hookup function is OK. I am using chan_h323 driver. I also tried changing codecs, g711u and g723.1. The result is same. My phones are no branded Taiwanese. I installed pwlib 1.5.0, openh323 1.12.0. H.323-phone is fastconnect mode. NetMeeting works both call. It's not using fastconnect mode. Could I have some suggestions? Regards, Shong Ching ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Chan_h323 one way audio .digium.com 18-08-03 12:19 PM Please respond to asterisk-users i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323.so native?
Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_oh323 Dial format / voice latency 4 to 5 secs
Hi, Can someone confirm the format of the Dial string for a H.323 gateway using chan_oh323? The format I have working is: exten = 5000,1,Dial(OH323/h323:[EMAIL PROTECTED]) I have 5000 as a speed dial - the extension functions, but the voice latency within the call to the analog phone (99451133) is about 4 -5 secs. There is no delay from the internal calling extension - SIP. Outbound latency is very bad. (SIP to SIP - perfect) The CiscoGW is a 2600 with FXO ports. I also see bad voice latency when using SJPhone (h.323) to asterisk - SIP 7960. Again the voice delay seems to be on the outgoing h.323 call from asterisk. Inbound is OK - this scenario seems to rule out any problems with the Cisco config. Thanks in advance for any wisdom. Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 gateway call lost after 74sec always
Hi Michael, I have just updated to 0.5.4 and the problem is still there. Are there any parameters or logs that I should be checking? When I run SJPhone (h323) direct to the Cisco 2600 fxo gateway - the call remians up without error. When I run SJPhone (h323) to Asterisk and then to a SIP extension the call also remains up without error. The issue only displays on outbound h323 connections from asterisk. Thanks for your help. Regards, Steve. Michael Manousos [EMAIL PROTECTED]To: [EMAIL PROTECTED] .com cc: Sent by: Subject: Re: [Asterisk-Users] h323 gateway call lost after 74sec always [EMAIL PROTECTED] .digium.com 25-07-03 12:33 AM Please respond to asterisk-users Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered SIP/6000-9794 15:20.606 H225 Caller:80eea08 H225Received connect PDU. CONTROL PROTOCOL ERROR * Roundtrip Delay * -- Hungup 'H323:29764' == Spawn extension (default, 5500, 1) exited non-zero on 'SIP/6000-9794' Any ideas? Thanks. Yes, try the 0.5.4 version of asterisk-oh323. It's been fixed. Steve. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 gateway call lost after 74sec always
Michael, my mistake - more testing confirmed that the wrapper did not update in the correct location. Asterisk was still using 0.5.3. Replaced with 0.5.4 and the call is no longer dropped. Asterisk still reports H.323 CONTROL PROTOCOL ERROR (Roundtrip Delay) but it keeps hold of the call. Thanks for you help. Regards, Steve. Michael Manousos [EMAIL PROTECTED]To: [EMAIL PROTECTED] .com cc: Sent by: Subject: Re: [Asterisk-Users] h323 gateway call lost after 74sec always [EMAIL PROTECTED] .digium.com 25-07-03 12:33 AM Please respond to asterisk-users Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered SIP/6000-9794 15:20.606 H225 Caller:80eea08 H225Received connect PDU. CONTROL PROTOCOL ERROR * Roundtrip Delay * -- Hungup 'H323:29764' == Spawn extension (default, 5500, 1) exited non-zero on 'SIP/6000-9794' Any ideas? Thanks. Yes, try the 0.5.4 version of asterisk-oh323. It's been fixed. Steve. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 gateway call lost after 74sec always
Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered SIP/6000-9794 15:20.606 H225 Caller:80eea08 H225Received connect PDU. CONTROL PROTOCOL ERROR * Roundtrip Delay * -- Hungup 'H323:29764' == Spawn extension (default, 5500, 1) exited non-zero on 'SIP/6000-9794' Any ideas? Thanks. Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
