| Summary: | ASTERISK-22686: Asterisk does not sends RTP when transfer is done in telco side | ||
| Reporter: | Dalius M. (mdalius) | Labels: | |
| Date Opened: | 2013-10-14 04:13:14 | Date Closed: | 2013-10-29 09:58:45 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | 1.8.23.1 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 2013-10-08_2.pcap | |
| Description: | Hello,
We have found strange Asterisk issue. Phone attached to Asterisk (A) calls to SIP provider (B). (B) Puts call on hold and does attended transfer to (C). Asterisk plays MOH to (A) phone. When transfer is done, SIP provider sends INVITE, but Asterisk does not strart to send RTP. SIP provider told us that it is Asterisk bug. I have attached dump file from PBX's side. There is 2 VoIP calls, one from my softphone (IP 78.59.84.76) to PBX (IP 82.135.219.194) and other from PBX to SIP provider (IP 85.206.138.84). I have also tried Asterisk 11.3 version and Cisco SPA 504G (firmware 7.5.5) phone with same results, transfered calls are dropped | ||
| Comments: | By: Dalius M. (mdalius) 2013-10-14 04:15:06.034-0500 Trace file. By: Matt Jordan (mjordan) 2013-10-28 12:18:27.745-0500 The IP addresses in your issue description don't match the IP addresses in the pcap. Can you provide what phone is what in the pcap, so bug marshals can analyze the issue easier? Additionally, please provide your sip.conf. By: Dalius M. (mdalius) 2013-10-29 09:12:37.358-0500 It is everything ok with IP addresses. PBX has 2 IP's: 82.135.219.194 other local and does not used in this call. My softphone has: 78.59.84.76 Provider has: 85.206.138.84 I have found that my issue is related to https://issues.asterisk.org/jira/browse/ASTERISK-20633 I have configured in sip.conf "ignoresdpversion=yes" and everything looks fine now. By: Matt Jordan (mjordan) 2013-10-29 09:58:45.937-0500 See ASTERISK-20633. | ||